In telecommunications networks, information is transferred in an encoded form between a transmitting communication device and a receiving communication device, such as an originating station and a terminating station. The transmitting communication device encodes original information, such as voice signals, into encoded information and sends it to the receiving communication device. The receiving communication device decodes the received encoded information to recreate the original information. The encoding and decoding is performed using codecs. The encoding of voice signals is performed in a codec located in the transmitting communication device, and the decoding is performed in a codec located in the receiving communication device.
There are many different speech codecs. With some codecs, transcoding may be required when the source and destination devices use incompatible codecs. Transcoding is a process by which a voice signal encoded according to one rate and encoding standard is converted to another encoding standard and possibly another rate. Transcoding can introduce latency and degradation in the voice signal being transmitted. To avoid the difficulties associated with transcoding, transcoder-free operation (TrFO) has been developed. With transcoder-free operation, a connection is established between telecommunications endpoints, such as mobile telephones and/or non-mobile telephones, that have compatible codecs so the connection does not use transcoders. TrFO has been widely deployed to eliminate the quality degradation due to network transcoding. Transcoding is not limited to the case when the source and the destination device use incompatible codecs. Additionally, tandem free operation (TFO) is a technique that may be used to deliver encoded information from one device to another device faithfully when the core network is circuit-switched.
Some telecommunications endpoints allow users to make calls over the Internet. Voice over Internet Protocol (VoIP) uses communication protocols and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. VoIP systems use session control protocols to control the set up and ending of calls, and use audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP. Often a range of codecs are used. Some implementations use narrowband and compressed speech, while others support high fidelity stereo codecs. VoIP systems do not use TFO or TrFO and may not perform transcoding.
Some codecs may not be implemented, however, because of infrastructure upgrade cost to operators and service providers as well as the core architectural complexity and availability of appropriate handsets. For example, Enhanced Variable Rate Wideband Codec (also referred to as EVRC-WB), developed with the latest speech compress technology, is able to carry speech characteristics from 50 Hz to 7 kHz and therefore provides clearer and more natural sounds than any of its predecessors in the EVRC (Enhanced Variable Rate Codec) family of codecs, like EVRC and EVRC-B which convey speech contents from 300 Hz to 4 kHz, that have been operated in cdma2000 networks. However, cdma2000 operators have not introduced it due to a lack of understanding of the infrastructure upgrade cost and the hardware modifications that would be required for handsets.